WO2006104017A1 - Dispositif de codage sonore et procédé de codage sonore - Google Patents

Dispositif de codage sonore et procédé de codage sonore Download PDF

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Publication number
WO2006104017A1
WO2006104017A1 PCT/JP2006/305871 JP2006305871W WO2006104017A1 WO 2006104017 A1 WO2006104017 A1 WO 2006104017A1 JP 2006305871 W JP2006305871 W JP 2006305871W WO 2006104017 A1 WO2006104017 A1 WO 2006104017A1
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Prior art keywords
amplitude ratio
quantization
delay difference
prediction parameter
signal
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PCT/JP2006/305871
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English (en)
Japanese (ja)
Inventor
Koji Yoshida
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Matsushita Electric Industrial Co., Ltd.
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Publication date
Application filed by Matsushita Electric Industrial Co., Ltd. filed Critical Matsushita Electric Industrial Co., Ltd.
Priority to JP2007510437A priority Critical patent/JP4887288B2/ja
Priority to ES06729819.0T priority patent/ES2623551T3/es
Priority to EP06729819.0A priority patent/EP1858006B1/fr
Priority to US11/909,556 priority patent/US8768691B2/en
Priority to CN2006800096953A priority patent/CN101147191B/zh
Publication of WO2006104017A1 publication Critical patent/WO2006104017A1/fr

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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/032—Quantisation or dequantisation of spectral components

Definitions

  • the present invention relates to a speech coding apparatus and speech coding method, and more particularly to a speech coding apparatus and speech coding method for stereo speech.
  • a voice coding scheme having a scalable configuration is desired in order to control traffic on the network and realize multicast communication.
  • a scalable configuration is a configuration in which speech data can be decoded even with a partial code and data power on the receiving side.
  • mono-rural stereo can be selected by allowing the receiving side to perform decoding of a stereo signal and decoding of a monaural signal using a part of the encoded data. Coding with a scalable configuration between them (monaural stereo 'scalable configuration) is desired.
  • a speech coding method having such a monaural, one-stereo, scalable configuration for example, prediction of signals between channels (hereinafter abbreviated as “ch” as appropriate) (from the 1st ch signal to the 2nd ch signal) Prediction or prediction from the 2nd channel signal to the 1st channel signal) is performed by pitch prediction between channels, that is, a code is performed using correlation between two channels (Non-Patent Document). 1).
  • Non-Special Terms 1 Ramprashad, SA, "Stereophonic CELP coding using cross channel p rediction", Proc. IEEE Workshop on Speech Coding, pp.136-138, Sep. 2000.
  • Non-Patent Document 1 the prediction parameters between channels (the delay and gain of pitch prediction between channels) are independently encoded, so The efficiency is not high.
  • An object of the present invention is to provide a speech encoding apparatus and speech encoding method that can efficiently encode stereo speech.
  • the speech coding apparatus includes a prediction parameter analysis unit that obtains a delay difference and an amplitude ratio between a first signal and a second signal as a prediction parameter, and between the delay difference and the amplitude ratio. And a quantization means for obtaining the prediction parameter force quantization prediction parameter based on the correlation.
  • stereo sound can be efficiently encoded.
  • FIG. 1 is a block diagram showing a configuration of a speech coding apparatus according to Embodiment 1.
  • FIG. 2 is a block diagram showing a configuration of a second channel prediction unit according to Embodiment 1
  • FIG. 3 is a block diagram showing a configuration of a prediction parameter quantization unit according to Embodiment 1 (Configuration Example 1).
  • FIG. 4 is a characteristic diagram showing an example of a prediction parameter codebook according to Embodiment 1
  • FIG. 5 is a block diagram showing a configuration of a prediction parameter quantization unit according to Embodiment 1 (Configuration Example 2).
  • FIG. 6 is a characteristic diagram showing an example of a function used in the amplitude ratio estimation unit according to Embodiment 1.
  • FIG. 7 is a block diagram showing a configuration of a prediction parameter quantization unit according to Embodiment 2 (configuration example 3).
  • FIG. 8 is a characteristic diagram showing an example of a function used in the distortion calculation unit according to the second embodiment.
  • FIG. 9 is a block diagram showing a configuration of a prediction parameter quantization unit according to Embodiment 2 (configuration example 4).
  • FIG. 10 is a characteristic diagram showing an example of functions used in the amplitude ratio correction unit and the amplitude ratio estimation unit according to Embodiment 2.
  • FIG. 11 is a block diagram showing a configuration of a prediction parameter quantization unit according to Embodiment 2 (configuration example 5).
  • the configuration of the speech coding apparatus according to this embodiment is shown in FIG.
  • the speech encoding apparatus 10 shown in FIG. 1 includes an lch encoding unit 11, an lch decoding unit 12, a 2ch prediction unit 13, a subtractor 14, and a second channel prediction residual code input unit 15.
  • the operation is assumed on a frame basis.
  • the code signal data (the l-th channel code data) of the audio signal is output to the l-channel decoder 12. Further, the lch code data is multiplexed with the 2ch prediction parameter encoded data and the 2ch encoded data and transmitted to a speech decoding apparatus (not shown).
  • the lch decoding unit 12 generates an lch decoded signal from the lch encoded data and outputs the lch decoded signal to the second channel prediction unit 13.
  • the second channel prediction parameter code data obtained by encoding the second channel prediction parameter is output.
  • the second channel prediction parameter encoded data is multiplexed with other encoded data and transmitted to a speech decoding apparatus (not shown).
  • Second channel prediction residual encoding section 15 encodes the second channel prediction residual signal and outputs second channel encoded data.
  • the second channel code data is multiplexed with other code data and transmitted to the audio decoder.
  • FIG. 2 shows the configuration of the second channel prediction unit 13.
  • the second channel prediction unit 13 includes a prediction parameter analysis unit 21, a prediction parameter quantization unit 22, and a signal prediction unit 23.
  • the second channel prediction unit 13 uses parameters based on the delay difference D and the amplitude ratio g of the second channel audio signal with respect to the lch audio signal based on the correlation between the channel signals of the stereo signal.
  • the 1st ch audio signal strength predicts the 2nd ch audio signal.
  • the prediction parameter analysis unit 21 obtains the delay difference D and the amplitude ratio g of the second channel audio signal with respect to the first channel audio signal from the first channel decoded signal and the second channel audio signal as inter-channel prediction parameters, and calculates the prediction parameter quantum. To the conversion unit 22.
  • the prediction parameter quantization unit 22 quantizes the input prediction parameter (delay difference D, amplitude ratio g), and outputs the quantized prediction parameter and the second channel prediction parameter encoded data.
  • the quantized prediction parameter is input to the signal prediction unit 23.
  • Prediction parameter quantization part 2
  • the signal prediction unit 23 performs prediction of the second channel signal using the l-th channel decoded signal and the quantized prediction parameter, and outputs the prediction signal.
  • sp_ch2 (n) g ⁇ sd_chl (n-D)... (1)
  • the prediction parameter analysis unit 21 minimizes the distortion Dist represented by Equation (2), that is, the distortion Dist between the second channel audio signal s_ch2 (n) and the second channel prediction signal sp_ch2 (n).
  • the prediction parameters (delay difference D, amplitude ratio g) are obtained.
  • the prediction parameter analysis unit 21 The delay difference D that maximizes the cross-correlation between the voice signal and the l-th channel decoded signal or the ratio g of the average amplitude per frame may be obtained as the prediction parameter.
  • the prediction parameter quantization unit 22 uses this relationship to efficiently encode the inter-channel prediction parameters (delay difference D, amplitude ratio g), and with a smaller number of quantization bits. Realize equivalent quantization distortion.
  • the configuration of the prediction parameter quantization unit 22 according to the present embodiment is as shown in Fig. 3 ⁇ Configuration example 1> or Fig. 5 Configuration example 2>.
  • the delay difference D and the amplitude ratio g are represented as a two-dimensional vector, and vector quantization is performed on the two-dimensional vector.
  • Figure 4 is a characteristic diagram of the symbol vector that represents this two-dimensional vector with a dot ( ⁇ ).
  • the distortion calculation unit 31 applies each code vector of the prediction parameter codebook 33 to a prediction parameter represented by a two-dimensional vector (D, g) that also has a delay difference D, an amplitude ratio g, and a force. Calculate the distortion between and.
  • the minimum distortion search unit 32 searches for the code vector having the smallest distortion among all the code vectors, sends the search result to the prediction parameter codebook 33, and supports the code vector.
  • the index to be output is output as the 2nd channel prediction parameter code key data.
  • the prediction parameter codebook 33 outputs the code vector as a quantized prediction parameter based on the search result! /, With the least distortion! /.
  • wd and wg are weight constants that adjust the weight between the quantization distortion for the delay difference at the time of distortion calculation and the quantization distortion for the amplitude ratio.
  • a function for estimating the amplitude ratio g from the delay difference D is determined in advance, and after the delay difference D is quantized, its quantized power is predicted residual with respect to the amplitude ratio estimated using that function.
  • the delay difference quantization unit 51 performs quantization on the delay difference D of the prediction parameters, outputs the quantization delay difference Dq to the amplitude ratio estimation unit 52, and Output as prediction parameter. Also, the delay difference quantization unit 51 outputs the quantized delay difference index obtained by quantizing the delay difference D as the second channel prediction parameter code key data.
  • the amplitude ratio estimator 52 calculates an amplitude ratio estimate (estimated amplitude ratio) gp from the quantization delay difference Dq, and outputs it to the amplitude ratio estimation residual quantizer 53.
  • a function for estimating the amplitude ratio is also used for the quantization delay differential force prepared in advance. This function is quantized
  • a plurality of data indicating the correspondence between the delay difference Dq and the estimated amplitude ratio gp is obtained from the stereo audio signal for learning, and the correspondence is also prepared in advance by learning.
  • Amplitude ratio estimation residual quantization section 53 obtains estimated residual ⁇ g with respect to estimated amplitude ratio gp of amplitude ratio g according to equation (4).
  • the amplitude ratio estimation residual quantization unit 53 quantizes the estimated residual ⁇ g obtained by Equation (4), and outputs the quantization estimated residual as a quantization prediction parameter. .
  • the amplitude ratio estimation residual quantization unit 53 outputs a quantization estimation residual index obtained by quantization of the estimation residual ⁇ g as second channel prediction parameter code key data.
  • FIG. 6 shows an example of a function used in the amplitude ratio estimation unit 52.
  • the input prediction parameters (D, g) are shown as points on the coordinate plane in Fig. 6 as 2D vectors.
  • the amplitude ratio estimation residual quantization unit 53 obtains an estimated residual ⁇ g with respect to the estimated amplitude ratio gp of the amplitude ratio g of the input prediction parameter, and quantizes the estimated residual ⁇ g.
  • the quantization error can be reduced as compared with the case where the amplitude ratio is directly quantized, and as a result, the quantization efficiency can be improved.
  • the estimated amplitude ratio gp is obtained from the quantized delay difference Dq using a function for estimating the quantized delay differential force amplitude ratio, and the input amplitude ratio g relative to the estimated amplitude ratio gp is calculated.
  • the configuration for quantizing the estimated residual ⁇ g has been described, but the estimated delay difference is calculated from the quantized amplitude ratio gq using a function for quantizing the input amplitude ratio g and estimating the delay difference from the quantized amplitude ratio.
  • a configuration may be adopted in which Dp is obtained and the estimated residual ⁇ D of the input delay difference D relative to the estimated delay difference Dp is quantized.
  • the speech coding apparatus is different from Embodiment 1 in the configuration of prediction parameter quantization section 22 (FIGS. 2, 3, and 5). Quantization of prediction parameters in this embodiment Then, in the quantization of the delay difference and the amplitude ratio, quantization is performed so that the quantization errors of both parameters are audibly cancelled. In other words, when the quantization error of the delay difference occurs in the positive direction, the quantization error of the amplitude ratio is quantified to be larger, and conversely, when the quantization error of the delay difference occurs in the negative direction. Quantization is performed so that the quantization error of the amplitude ratio becomes smaller.
  • the delay difference and the amplitude ratio are quantized by adjusting the quantization error of the delay difference and the quantization error of the amplitude ratio so that the stereo localization does not change audibly.
  • the prediction parameter can be encoded more efficiently. That is, equivalent sound quality can be achieved at a lower code bit rate or higher sound quality at the same code bit rate.
  • the configuration of the prediction parameter quantization unit 22 according to the present embodiment is as shown in Fig. 7 ⁇ Configuration Example 3> or Fig. 9 Configuration Example 4>.
  • Configuration Example 3 differs from Configuration Example 1 (Fig. 3) in calculating the distortion.
  • FIG. 7 the same components as those in FIG.
  • the distortion calculation unit 71 performs each code vector of the prediction parameter codebook 33 on the prediction parameter represented by a two-dimensional vector (D, g) composed of the delay difference D and the amplitude ratio g. Calculate the distortion between.
  • the distortion calculation unit 71 is input.
  • the two-dimensional beta (D, g) of the prediction parameter is the nearest hearing-equivalent point (Dc '(k), gc' () to each code vector (Dc (k), gc (k))
  • w d and wg are weight constants for adjusting the weight between the quantization distortion for the delay difference at the time of distortion calculation and the quantization distortion for the amplitude ratio.
  • Dst (k) wd-((Dc '(k) -Dc (k)) 2 + wg * (gc, (k) -gc (k)) 2 ... (5)
  • the closest auditory equivalent point to each code vector (Dc (k), gc (k)) is, as shown in FIG. 8, from each code vector, the input prediction parameter vector (D , g) and stereo orientation correspond to the point where the perpendicular line is dropped to function 81 which is audibly equivalent.
  • This function 81 is a function in which the delay difference D and the amplitude ratio g are proportional to the positive direction. The larger the delay difference is, the larger the amplitude ratio is. On the other hand, the smaller the delay difference is, the smaller the amplitude ratio is. This is based on the audible characteristic, which gives an audible equivalent orientation.
  • the input prediction parameter vector is (D, g) as perceptually closest to each code vector (Dc (k), gc (k)) (ie, on the vertical line) on the function 81.
  • Dc code vector
  • gc gc vector
  • the code vector A quantization distortion A
  • the code vector B In the case of quantization distortion (B), the code vector C (quantization distortion C), which is closer to the auditory sense of stereo localization than the input prediction parameter vector, becomes the quantized value, and quantization can be performed with less auditory distortion.
  • the delay difference quantization unit 51 also outputs the quantization delay difference Dq to the amplitude ratio correction unit 91.
  • the amplitude ratio correction unit 91 corrects the amplitude ratio g to an audibly equivalent value based on the quantization error of the delay difference to obtain a corrected amplitude ratio g '.
  • the corrected amplitude ratio g ′ is input to the amplitude ratio estimation residual quantization unit 92.
  • Amplitude ratio estimation residual quantization section 92 obtains an estimation residual Sg with respect to estimated amplitude ratio gp of corrected amplitude ratio g 'according to equation (6).
  • the amplitude ratio estimation residual quantization unit 92 quantizes the estimated residual ⁇ g obtained by Equation (6), and outputs the quantization estimated residual as a quantization prediction parameter. . Further, the amplitude ratio estimation residual quantization unit 92 outputs a quantization estimation residual index obtained by quantization of the estimation residual ⁇ g as the second channel prediction parameter code key data.
  • FIG. 10 shows an example of functions used in the amplitude ratio correction unit 91 and the amplitude ratio estimation unit 52.
  • the function 81 used in the amplitude ratio correction unit 91 is the same function as the function 81 used in the configuration example 3, and the function 61 used in the amplitude ratio estimation unit 52 is used in the configuration example 2. This is the same function as function 61.
  • the function 81 is a function in which the delay difference D and the amplitude ratio g are proportional to the positive direction as described above, and the amplitude ratio correction unit 91 uses the function 81 to quantize the delay difference Dq.
  • a corrected amplitude ratio g ′ audibly equivalent to the amplitude ratio g based on the quantization error of the delay difference is obtained.
  • the amplitude ratio estimated residual quantization unit 92 obtains an estimated residual ⁇ g with respect to the estimated amplitude ratio gp of the corrected amplitude ratio g ′, and quantizes the estimated residual ⁇ g.
  • the estimated residual is obtained from the amplitude ratio (corrected amplitude ratio) corrected to an audibly equivalent value based on the quantization error of the delay difference, and the estimated residual is quantized. Quantization with small distortion and small quantization error can be performed.
  • the auditory characteristics relating to the delay difference and the amplitude ratio may be used as in the present embodiment.
  • the configuration of the prediction parameter quantization unit 22 in this case is as shown in FIG. In FIG. 11, the same components as those in Configuration Example 4 (FIG. 9) are denoted by the same reference numerals.
  • the amplitude ratio correction unit 91 corrects the amplitude ratio g to an audibly equivalent value based on the quantization error of the delay difference to obtain the corrected amplitude ratio g ′. .
  • the corrected amplitude ratio g ′ is input to the amplitude ratio quantization unit 1101. [0059]
  • the amplitude ratio quantization unit 1101 quantizes the corrected amplitude ratio g 'and outputs the quantized amplitude ratio as a quantization prediction parameter.
  • amplitude ratio quantization section 1101 outputs a quantized amplitude ratio index obtained by quantization of corrected amplitude ratio g ′ as second channel prediction parameter code data.
  • the prediction parameters (delay difference D and amplitude ratio g) have been described as scalar values (one-dimensional values), but over a plurality of time units (frames). Quantization similar to the above may be performed by combining a plurality of obtained prediction parameters into a vector of two or more dimensions.
  • each of the above embodiments can also be applied to a voice codec apparatus having a monaural-stereo's scalable configuration.
  • a monaural signal is generated from the input stereo signal (the lch and 2ch audio signals) and encoded
  • the lch is obtained from the monaural decoded signal by inter-channel prediction.
  • a (or 2nd ch) speech signal is predicted, and a prediction residual signal between this prediction signal and the lch (or 2nd ch) speech signal is encoded.
  • CELP codes are used for the monaural core layer and stereo enhancement layer codes, and inter-channel prediction is performed on the monaural driving sound source signal obtained by the mono core layer in the stereo enhancement layer. May be encoded with the CELP excitation code.
  • the inter-channel prediction parameter is a parameter for predicting the lch (or 2nd ch) speech signal with monaural signal power.
  • the delay difference Dml, Dm2 the amplitude ratio of the 1st channel audio signal and the 2nd channel audio signal relative to the monaural signal gml and gm2 may be combined for two channel signals and quantized in the same manner as in the second embodiment.
  • the speech coding apparatus is used in a mobile communication system! Installed in wireless communication devices such as wireless communication mobile station devices and wireless communication base station devices It is also possible to do.
  • Each functional block used in the description of each of the above embodiments is typically realized as an LSI which is an integrated circuit. These may be individually made into one chip, or may be made into one chip so as to include a part or all of them.
  • the method of circuit integration is not limited to LSI, and may be realized by a dedicated circuit or a general-purpose processor. You may use an FPGA (Field Programmable Gate Array) that can be programmed after manufacturing the LSI, or a reconfigurable processor that can reconfigure the connection and settings of the circuit cells inside the LSI.
  • FPGA Field Programmable Gate Array
  • the present invention can be applied to the use of a communication apparatus in a mobile communication system, a packet communication system using the Internet protocol, or the like.

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  • Computational Linguistics (AREA)
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  • Audiology, Speech & Language Pathology (AREA)
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Abstract

L’invention concerne un dispositif de codage sonore permettant de coder de manière efficace un son stéréophonique. Dans ce dispositif de codage sonore, une section d’analyse de paramètres de prédiction (21) détermine la différence de temporisation D et le rapport d’amplitude g d’un signal sonore de premier canal par rapport à un signal sonore de second canal comme paramètres de prédiction d’un canal à l’autre, à partir d’un signal décodé de premier canal et d’un signal sonore de second canal, une section de quantification de paramètres de prédiction (22) quantifie les paramètres de prédiction, une section de prédiction de signal (23) prédit un signal de second canal à l’aide du premier signal décodé et des paramètres de prédiction de quantification. La section de quantification de paramètres de prédiction (22) code et quantifie les paramètres de prédiction (la différence de temporisation D et le rapport d’amplitude g) à l’aide de la relation (corrélation) entre la différence de temporisation D et le rapport d’amplitude g attribuée à la caractéristique spatiale (par exemple la distance) entre la source sonore du signal et le point de réception.
PCT/JP2006/305871 2005-03-25 2006-03-23 Dispositif de codage sonore et procédé de codage sonore WO2006104017A1 (fr)

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Application Number Priority Date Filing Date Title
JP2007510437A JP4887288B2 (ja) 2005-03-25 2006-03-23 音声符号化装置および音声符号化方法
ES06729819.0T ES2623551T3 (es) 2005-03-25 2006-03-23 Dispositivo de codificación de sonido y procedimiento de codificación de sonido
EP06729819.0A EP1858006B1 (fr) 2005-03-25 2006-03-23 Dispositif de codage sonore et procédé de codage sonore
US11/909,556 US8768691B2 (en) 2005-03-25 2006-03-23 Sound encoding device and sound encoding method
CN2006800096953A CN101147191B (zh) 2005-03-25 2006-03-23 语音编码装置和语音编码方法

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JP2005-088808 2005-03-25

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EP (1) EP1858006B1 (fr)
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CN (1) CN101147191B (fr)
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JPWO2006104017A1 (ja) 2008-09-04
EP1858006A4 (fr) 2011-01-26
CN101147191A (zh) 2008-03-19
US8768691B2 (en) 2014-07-01
EP1858006B1 (fr) 2017-01-25
EP1858006A1 (fr) 2007-11-21
JP4887288B2 (ja) 2012-02-29
US20090055172A1 (en) 2009-02-26
ES2623551T3 (es) 2017-07-11
CN101147191B (zh) 2011-07-13

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